Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
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modules/video_coding/packet_buffer.h
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modules/video_coding/packet_buffer.h
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/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_VIDEO_CODING_PACKET_BUFFER_H_
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#define WEBRTC_MODULES_VIDEO_CODING_PACKET_BUFFER_H_
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#include <memory>
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#include <set>
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#include <vector>
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#include "webrtc/modules/include/module_common_types.h"
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#include "webrtc/modules/video_coding/packet.h"
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#include "webrtc/modules/video_coding/rtp_frame_reference_finder.h"
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#include "webrtc/modules/video_coding/sequence_number_util.h"
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#include "webrtc/rtc_base/criticalsection.h"
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#include "webrtc/rtc_base/scoped_ref_ptr.h"
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#include "webrtc/rtc_base/thread_annotations.h"
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namespace webrtc {
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class Clock;
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namespace video_coding {
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class FrameObject;
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class RtpFrameObject;
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// A received frame is a frame which has received all its packets.
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class OnReceivedFrameCallback {
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public:
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virtual ~OnReceivedFrameCallback() {}
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virtual void OnReceivedFrame(std::unique_ptr<RtpFrameObject> frame) = 0;
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};
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class PacketBuffer {
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public:
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static rtc::scoped_refptr<PacketBuffer> Create(
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Clock* clock,
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size_t start_buffer_size,
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size_t max_buffer_size,
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OnReceivedFrameCallback* frame_callback);
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virtual ~PacketBuffer();
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// Returns true if |packet| is inserted into the packet buffer, false
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// otherwise. The PacketBuffer will always take ownership of the
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// |packet.dataPtr| when this function is called. Made virtual for testing.
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virtual bool InsertPacket(VCMPacket* packet);
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void ClearTo(uint16_t seq_num);
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void Clear();
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void PaddingReceived(uint16_t seq_num);
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// Timestamp (not RTP timestamp) of the last received packet/keyframe packet.
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rtc::Optional<int64_t> LastReceivedPacketMs() const;
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rtc::Optional<int64_t> LastReceivedKeyframePacketMs() const;
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int AddRef() const;
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int Release() const;
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protected:
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// Both |start_buffer_size| and |max_buffer_size| must be a power of 2.
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PacketBuffer(Clock* clock,
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size_t start_buffer_size,
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size_t max_buffer_size,
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OnReceivedFrameCallback* frame_callback);
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private:
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friend RtpFrameObject;
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// Since we want the packet buffer to be as packet type agnostic
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// as possible we extract only the information needed in order
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// to determine whether a sequence of packets is continuous or not.
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struct ContinuityInfo {
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// The sequence number of the packet.
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uint16_t seq_num = 0;
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// If this is the first packet of the frame.
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bool frame_begin = false;
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// If this is the last packet of the frame.
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bool frame_end = false;
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// If this slot is currently used.
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bool used = false;
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// If all its previous packets have been inserted into the packet buffer.
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bool continuous = false;
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// If this packet has been used to create a frame already.
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bool frame_created = false;
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};
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Clock* const clock_;
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// Tries to expand the buffer.
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bool ExpandBufferSize() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
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// Test if all previous packets has arrived for the given sequence number.
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bool PotentialNewFrame(uint16_t seq_num) const
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RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
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// Test if all packets of a frame has arrived, and if so, creates a frame.
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// Returns a vector of received frames.
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std::vector<std::unique_ptr<RtpFrameObject>> FindFrames(uint16_t seq_num)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
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// Copy the bitstream for |frame| to |destination|.
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// Virtual for testing.
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virtual bool GetBitstream(const RtpFrameObject& frame, uint8_t* destination);
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// Get the packet with sequence number |seq_num|.
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// Virtual for testing.
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virtual VCMPacket* GetPacket(uint16_t seq_num)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
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// Mark all slots used by |frame| as not used.
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// Virtual for testing.
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virtual void ReturnFrame(RtpFrameObject* frame);
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void UpdateMissingPackets(uint16_t seq_num)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
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rtc::CriticalSection crit_;
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// Buffer size_ and max_size_ must always be a power of two.
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size_t size_ RTC_GUARDED_BY(crit_);
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const size_t max_size_;
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// The fist sequence number currently in the buffer.
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uint16_t first_seq_num_ RTC_GUARDED_BY(crit_);
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// If the packet buffer has received its first packet.
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bool first_packet_received_ RTC_GUARDED_BY(crit_);
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// If the buffer is cleared to |first_seq_num_|.
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bool is_cleared_to_first_seq_num_ RTC_GUARDED_BY(crit_);
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// Buffer that holds the inserted packets.
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std::vector<VCMPacket> data_buffer_ RTC_GUARDED_BY(crit_);
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// Buffer that holds the information about which slot that is currently in use
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// and information needed to determine the continuity between packets.
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std::vector<ContinuityInfo> sequence_buffer_ RTC_GUARDED_BY(crit_);
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// Called when a received frame is found.
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OnReceivedFrameCallback* const received_frame_callback_;
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// Timestamp (not RTP timestamp) of the last received packet/keyframe packet.
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rtc::Optional<int64_t> last_received_packet_ms_ RTC_GUARDED_BY(crit_);
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rtc::Optional<int64_t> last_received_keyframe_packet_ms_
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RTC_GUARDED_BY(crit_);
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rtc::Optional<uint16_t> newest_inserted_seq_num_ RTC_GUARDED_BY(crit_);
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std::set<uint16_t, DescendingSeqNumComp<uint16_t>> missing_packets_
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RTC_GUARDED_BY(crit_);
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mutable volatile int ref_count_ = 0;
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};
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} // namespace video_coding
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} // namespace webrtc
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#endif // WEBRTC_MODULES_VIDEO_CODING_PACKET_BUFFER_H_
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